Massachusetts Institute of Technology Department of Electrical Engineering & Computer Science 6.

341: Discrete-Time Signal Processing Fall 2005

Project Assignment

Issued: Sept. 27, 2005

Project I due: Nov. 1, 2005

Project II due: Dec. 6, 2005

This handout contains all parts of both projects (Project I and Project II) to be assigned in 6.341 this semester. The writeup for Project I is due on November 1, 2005. The writeup for Project II is due on December 6, 2005. Both projects are required to be done, turned in by all students independent of which homework option you chose, and will be counted in the nal grade. You are required to turn in written reports for both Projects I and II. Project I will have two separate writeups, one for Part A and one for Part B. Project II will have one writeup. Each writeup should have a summary of how you went about it (anywhere from one to ve pages, not to exceed ve pages). In addition to the ve page maximum, you can include any appropriate Matlab plots. Do NOT turn in any Matlab code. The total number of pages including description and plots for Project I Part A should not exceed ve pages and for Project I part B should not exceed 10 pages. The total maximum number of pages for Project II is 10 pages. No formal solutions to the project will be handed out. We may select one or two of the reports handed in and (with the permission of the student) will distribute those to the entire class. Each writeup (two for Project I, one for Project II) will receive a grade which reects our assessment of your level of understanding of the various issues involved and the eort that you put into it. For the actual grading, we will be using the EGRMU grading scheme which is an acronym for E(xtraordinary), G(ood), R(easonable, kind of), M(arginal), U(nsatisfactory). We are likely to use some mixed grades like E/G etc. In generating a numerical grade at the end of the semester, we will use the following scale: E=100, E/G=90, G=80, G/R=70, R=60, R/M=50, M=30, M/U=15, U=0. COMPLETION OF THE PROJECTS AND WRITEUPS IS A REQUIREMENT TO PASS THE COURSE.

Note: Matlab help sessions will be held by the TAs to help you get started on the project. This means that the TAs will be available in a cluster and will be open to questions. These help sessions are intended for Matlab- and project-related questions and are not to be used as general oce hours. Also note that, to encourage you to start early on the project, the last problem in problem set 5 asks about your progress on the project.

Playing Sound in Matlab

Throughout these projects it will be necessary to play audio les which you will have manipulated in Matlab. While the function sound works well for doing this on Mac (OS X) and PC platforms, using sound on Athena machines is problematic and should be avoided. The sta has instead written several functions which should be used to play sound in Matlab on Athena Sun (UNIX) machines in lieu of sound. (Athena Linux-based machines are not supported within the context of this project.) These functions, usound441 and usound110, are available on the course website as .m les and will play back sequences at sampling rates of 44100 Hz and 11025 Hz, respectively. To install, simply put usound441.m and usound110.m in a directory which is in your Matlab path, or in your working directory. For information on how to use the functions type help usound441 or help usound110. Note that you may need to type add infoagents at your Athena prompt before starting Matlab for these functions to work properly.

Overview
One area in which discrete-time signal processing has enjoyed widespread use is in the eld of speech processing. In this set of projects we consider various issues relating to applying digital ltering to speech. Specically, in the rst part of Project I, we consider the eect of all-pass ltering on speech. Your current background in 6.341 should be sucient to allow you to begin on this part immediately. In the second part of Project I we consider IIR and FIR lter design in the context of enhancing speech corrupted by additive noise. While this relies somewhat on material in Lectures 7-8, some aspects can be started earlier. In Project II we continue the second part of Project I, with a constraint imposed on the allowable multiplier rate (number of multiplies per input sample) and the eect of this on dierent lter designs. Although Project II isnt due until December 6, it does not rely on lectures beyond L10 and we concourage you to start on it as soon as possible. Historically, DSP courses have put considerable emphasis on design techniques for both IIR and FIR lters, as discussed in Chapter 7 of the course text. A variety of lter design algorithms are now implemented in common software packages such as Matlab. This makes it unnecessary for most practitioners to learn the details of the design algorithms; however, ceding the design function to a software package makes it more important to understand the properties of dierent types of

optimal lters, the meanings of the design parameters, and some of the trade-os between lter classes. In the context of what we are covering this semester on IIR lter design, we suggest that you read Section 7.0 including Example 7.1. Also, read the examples in Section 7.1.3, but focus on what Butterworth and Chebyshev lters are rather than on the details of the bilinear transformation. For the discussion of FIR lter design, the suggested reading is Sections 7.2, 7.3, and 7.4 through 7.4.1. We will not be going into the details of the issues raised in Section 7.4.1, but we would recommend at least reading that section to get a sense of what some of those issues are.

Project I: (Due November 1, 2005)

Project I. A.
A common folk theorem states that the ear is insensitive to phase, i.e. that for audio, phase distortion is inaudible. If that is correct, then processing audio with an all-pass lter should not result in perceived distortion. This part of the project tests this conjecture. The all-pass lter that we consider is of the form
1 e ) (z 1 e k k )(z H (z ) = 1 )(1 e z 1 ) (1 e z k k k =1 3 N

6 k N k =0 bk z 6 k k =0 ak z

For this part of the project, the le projIA.mat will need to be rst loaded into Matlab. After downloading this le from the course website, copy it into your Matlab working directory and type load projIA. The bk and ak coecients above are stored in the variables b and a respectively; b(k + 1)= bk and a(k + 1)= ak . (a) Implement the all-pass lter for N = 1 and show plots of the impulse response (the rst 100 samples), the magnitude of the frequency response, and the group delay. (b) Plot the pole-zero diagram. You may nd the function roots helpful in doing so. (b) Process the speech le, stored as the variable speech (sampled at rate fs = 11025 Hz), using the command filter to run the lter implemented in (a). Listen to the result and comment specically on whether there is audible distortion. (c) Implement the lter for N = 50 and again show plots of the impulse response (the rst 5000 samples), the magnitude of the frequency response, and the group delay. (d) Repeat (b). In your writeup, comment specically on how you would subjectively describe the distortion and on what aspect of the lter is its primary cause.

Project I. B.
For this part of the project, the le projIB.mat will need to be rst loaded into Matlab. After downloading this le from the course website, copy it into your Matlab working directory and type load projIB. In this part of the project, it may be helpful to use Matlabs order estimation functions (e.g. buttord, cheb1ord, ...). A caveat in the use of these functions is that Matlabs denition of ripple diers between the IIR and FIR lter design functions. The following can be used to sort through these dierences in convention: 1. In the design of IIR lters, Matlab takes specications for ripple magnitude and stopband attenuation as input parameters. A specication of ripple magnitude IIR yields an IIR lter design with a maximum passband gain of unity (or 0 dB) and a minimum passband gain of 10IIR /20 (or IIR dB). A specication of stopband attenuation IIR yields an IIR lter design with a maximum stopband gain of 10IIR /20 (or IIR dB). 2. In the design of FIR lters, Matlab also takes a specication of ripple magnitude as an input parameter. A specication of ripple magnitude FIR yields a FIR lter design with a maximum passband gain of (1 + FIR ) (or 20 log10 (1 + FIR ) dB), and a minimum passband gain of (1 FIR ) (or 20 log 10 (1 FIR ) dB). A specication of stopband gain FIR yields an FIR lter design with a maximum stopband gain of FIR (or 20 log10 (FIR ) dB). Clearly, to compare FIR and IIR lters designed with Matlab, we must nd a correspondence between our desired specications and the input parameters to the lter design functions. Let us thus dene Gpbmax (dB), Gpbmin (dB), and Gsbmax (dB) to refer to the desired maximum passband gain, minimum passband gain, and maximum stopband gain, respectively. In order to have free control over each of these parameters, we also introuce kFIR and kIIR , linear scaling terms which are used to normalize our FIR and IIR designs. Before beginning Project I.B. do (a) and (b) below and include your solutions as part of the project writeup. (a) Find FIR , FIR , and kFIR in terms of Gpbmax , Gpbmin , and Gsbmax . These parameters will be used for FIR designs. (b) Find IIR , IIR , and kIIR in terms of Gpbmax , Gpbmin , and Gsbmax . These parameters will be used for IIR designs. Project I.B. is concerned with designing a low-pass lter for the removal of high-pass noise from a speech signal. The noisy signal is stored in the variable noisy and was sampled at 44100 (fs) Hz. It consists of a summation of a speech signal which was band-limited at 4 kHz using a low-pass lter with a very narrow transition band, and a noise signal which was ltered at 4 kHz using a high-pass lter with a very narrow transition band. To lter the noise we must design a discrete-time lter with the following parameters: 1. Passband edge: 2500 Hz. 2. Stopband edge: 4000 Hz. 4

3. Maximum gain in the passband Gpbmax : 40 dB. 4. Minimum gain in the passband Gpbmin : 37 dB. 5. Maximum gain in the stopband Gsbmax : 55 dB. Because it may take a few iterations to get each lter right, we suggest you write a .m Matlab script for each lter. Note that in this part of the project, you may end up with very high order lters since the specications are rather severe. You can try to implement the IIR lters according to the specs using Matlabs built in tools (such as the fdatool or the command line tools butter, cheby1, cheby2, and elliptype help signal for more information). If you do so, you will notice that the resulting systems might not be stable. There are several reasons for this, the most important of which being coecient quantizationeven with oating point precision. This issue arises because the specications are very tight and some of the lter types have all of their poles concentrated near z = 1. Fortunately, there is a workaround. What you need to do is to group the poles and zeros for the desired lter in pairs (conjugate or not) to create smaller stable second order lters of the form N (z )/D(z ) where N (z ) and D(z ) are at most second order polynomials in z . A cascade of such lters will produce the desired system, and Matlab will be able to analyze it. The drawback is that you will have to implement your own methods to generate the plots and simulate the system. Design a DT lter of each of the following types based on the specications given above: Butterworth, Chebyshev Type I, Chebyshev Type II, Elliptic, Parks-McClellan, Kaiser. For each of the designs, provide the following in your Part I project write-up: (a) The order of the lter. (b) The number of multiplication operations per input sample required to implement the lter. Be sure to explain the structure you assume. (c) Plot the magnitude response (in dB) from = 0 to = using freqz. Plot a detail of the magnitude response, focusing on the passband ripple (linear scale). Plot the group delay (in samples) using grpdelay. (Use subplot to t the three plots on the same page for each lter.) (d) Plot the pole-zero diagram. (e) Plot the impulse response using filter and stem for 100 samples. (Use subplot to t the pole-zero diagram and the impulse response on the same page.) Filter noisy using your de-noising lter. Listen to the ltered and original les. How do they compare?

Project II: (Due December 6, 2005)

In this project, you will design lters for a hardware implementation of the de-noising system in Project I B. Since the de-noising lter does a good job of keeping the resulting signal band-limited to 4 kHz, it is prudent to add a compressor-by-4 (which for no aliasing requires its input to be band-limited to 5512.5 Hz) as a nal stage in the implementation of our system. The sampling rate at the output of the system is thus 11025 Hz, and the sampling rate at the systems input is still 44100 Hz. The overall de-noising system, which operates on a noisy signal x to yield a cleaned signal x , is depicted in its nominal form below.

The hardware on which we would like to implement the de-noiser is indeed very limited; a maximum of 17 multiplications can be performed per input sample. (The 18th multiplier is used elsewhere to implement a digital volume control.) In an attempt to lessen the constraints on our design problem, we are told that we must adhere to all of the parameters outlined in Part I.B. except passband edge, which should be kept as close to the stopband edge as possible, while still meeting the multiplication constraint. Our new design criteria become: 1. Maximum number of multiplications per input sample: 17 2. Passband edge: as close to 4000 Hz as possible. 3. Stopband edge: 4000 Hz. 4. Maximum gain in the passband Gpbmax : 40 dB. 5. Minimum gain in the passband Gpbmin : 37 dB. 6. Maximum gain in the stopband Gsbmax : 55 dB. Again design a DT lter of each of the following types based upon these new specications: Butterworth, Chebyshev Type I, Chebyshev Type II, Elliptic, Parks-McClellan, Kaiser. Note that for this section, writing a .m script will almost certainly prove useful. For each of the designs, provide the following in your Project II project write-up: (a) The order of the lter. (b) The passband edge parameter chosen. (b) The number of required multiplications per input sample, both with and without the use of polyphase. Were there lter implementations which did not meet the required specication for any arbitrarily low choice of passband edge? (c) Plot the magnitude response (in dB) from = 0 to = using freqz. 6

Now implement your de-noising system using each lter you designed. For designs where a polyphase implementation is assumed for purposes of multiplication counting, feel free to do the actual ltering in Matlab on the high-rate side of the compressor, remembering that a polyphase implementation implies only a specic lter form and will therefore not change the overall behavior of the system. To implement a compressor-by-4 on a sequence x in Matlab, use x = x(1:4:length(x));. On the course website you will nd a list of unique .wav les which contain speech utterances in the presence of noise. Download the le corresponding to your Athena username and run it through the de-noising system you implemented, using each lter you designed. (You can read .wav les into Matlab using wavread.) In your writeup, specically comment on the following: How does the quality compare for each lter design? What aspects of each lter do you think contribute to this? What does your utterance say? Feel free to experiment with other users utterances as well.